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FFmpeg/libavcodec/aacenc.c
Rostislav Pehlivanov 0cf6853804 aacenc: quit when the audio queue reaches 0 rather than keeping track of empty frames 
The libopus encoder does the same thing and its better than
keeping track of when the empty flush frames appear.

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2016-11-08 00:50:51 +00:00

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/*
* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC encoder
*/
/***********************************
* TODOs:
* add sane pulse detection
***********************************/
#include "libavutil/libm.h"
#include "libavutil/thread.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
#include "internal.h"
#include "mpeg4audio.h"
#include "kbdwin.h"
#include "sinewin.h"
#include "aac.h"
#include "aactab.h"
#include "aacenc.h"
#include "aacenctab.h"
#include "aacenc_utils.h"
#include "psymodel.h"
static AVOnce aac_table_init = AV_ONCE_INIT;
/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
*/
static void put_audio_specific_config(AVCodecContext *avctx)
{
PutBitContext pb;
AACEncContext *s = avctx->priv_data;
int channels = s->channels - (s->channels == 8 ? 1 : 0);
init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
put_bits(&pb, 5, s->profile+1); //profile
put_bits(&pb, 4, s->samplerate_index); //sample rate index
put_bits(&pb, 4, channels);
//GASpecificConfig
put_bits(&pb, 1, 0); //frame length - 1024 samples
put_bits(&pb, 1, 0); //does not depend on core coder
put_bits(&pb, 1, 0); //is not extension
//Explicitly Mark SBR absent
put_bits(&pb, 11, 0x2b7); //sync extension
put_bits(&pb, 5, AOT_SBR);
put_bits(&pb, 1, 0);
flush_put_bits(&pb);
}
void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
{
++s->quantize_band_cost_cache_generation;
if (s->quantize_band_cost_cache_generation == 0) {
memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
s->quantize_band_cost_cache_generation = 1;
}
}
#define WINDOW_FUNC(type) \
static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
SingleChannelElement *sce, \
const float *audio)
WINDOW_FUNC(only_long)
{
const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
float *out = sce->ret_buf;
fdsp->vector_fmul (out, audio, lwindow, 1024);
fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
}
WINDOW_FUNC(long_start)
{
const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
float *out = sce->ret_buf;
fdsp->vector_fmul(out, audio, lwindow, 1024);
memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
}
WINDOW_FUNC(long_stop)
{
const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
float *out = sce->ret_buf;
memset(out, 0, sizeof(out[0]) * 448);
fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
}
WINDOW_FUNC(eight_short)
{
const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
const float *in = audio + 448;
float *out = sce->ret_buf;
int w;
for (w = 0; w < 8; w++) {
fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
out += 128;
in += 128;
fdsp->vector_fmul_reverse(out, in, swindow, 128);
out += 128;
}
}
static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
SingleChannelElement *sce,
const float *audio) = {
[ONLY_LONG_SEQUENCE] = apply_only_long_window,
[LONG_START_SEQUENCE] = apply_long_start_window,
[EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
[LONG_STOP_SEQUENCE] = apply_long_stop_window
};
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
float *audio)
{
int i;
const float *output = sce->ret_buf;
apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
else
for (i = 0; i < 1024; i += 128)
s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
}
/**
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
int w;
put_bits(&s->pb, 1, 0); // ics_reserved bit
put_bits(&s->pb, 2, info->window_sequence[0]);
put_bits(&s->pb, 1, info->use_kb_window[0]);
if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
put_bits(&s->pb, 6, info->max_sfb);
put_bits(&s->pb, 1, !!info->predictor_present);
} else {
put_bits(&s->pb, 4, info->max_sfb);
for (w = 1; w < 8; w++)
put_bits(&s->pb, 1, !info->group_len[w]);
}
}
/**
* Encode MS data.
* @see 4.6.8.1 "Joint Coding - M/S Stereo"
*/
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
{
int i, w;
put_bits(pb, 2, cpe->ms_mode);
if (cpe->ms_mode == 1)
for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
}
/**
* Produce integer coefficients from scalefactors provided by the model.
*/
static void adjust_frame_information(ChannelElement *cpe, int chans)
{
int i, w, w2, g, ch;
int maxsfb, cmaxsfb;
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
maxsfb = 0;
cpe->ch[ch].pulse.num_pulse = 0;
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
for (w2 = 0; w2 < ics->group_len[w]; w2++) {
for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
;
maxsfb = FFMAX(maxsfb, cmaxsfb);
}
}
ics->max_sfb = maxsfb;
//adjust zero bands for window groups
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
for (g = 0; g < ics->max_sfb; g++) {
i = 1;
for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
if (!cpe->ch[ch].zeroes[w2*16 + g]) {
i = 0;
break;
}
}
cpe->ch[ch].zeroes[w*16 + g] = i;
}
}
}
if (chans > 1 && cpe->common_window) {
IndividualChannelStream *ics0 = &cpe->ch[0].ics;
IndividualChannelStream *ics1 = &cpe->ch[1].ics;
int msc = 0;
ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
ics1->max_sfb = ics0->max_sfb;
for (w = 0; w < ics0->num_windows*16; w += 16)
for (i = 0; i < ics0->max_sfb; i++)
if (cpe->ms_mask[w+i])
msc++;
if (msc == 0 || ics0->max_sfb == 0)
cpe->ms_mode = 0;
else
cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
}
}
static void apply_intensity_stereo(ChannelElement *cpe)
{
int w, w2, g, i;
IndividualChannelStream *ics = &cpe->ch[0].ics;
if (!cpe->common_window)
return;
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
for (w2 = 0; w2 < ics->group_len[w]; w2++) {
int start = (w+w2) * 128;
for (g = 0; g < ics->num_swb; g++) {
int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
float scale = cpe->ch[0].is_ener[w*16+g];
if (!cpe->is_mask[w*16 + g]) {
start += ics->swb_sizes[g];
continue;
}
if (cpe->ms_mask[w*16 + g])
p *= -1;
for (i = 0; i < ics->swb_sizes[g]; i++) {
float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
cpe->ch[0].coeffs[start+i] = sum;
cpe->ch[1].coeffs[start+i] = 0.0f;
}
start += ics->swb_sizes[g];
}
}
}
}
static void apply_mid_side_stereo(ChannelElement *cpe)
{
int w, w2, g, i;
IndividualChannelStream *ics = &cpe->ch[0].ics;
if (!cpe->common_window)
return;
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
for (w2 = 0; w2 < ics->group_len[w]; w2++) {
int start = (w+w2) * 128;
for (g = 0; g < ics->num_swb; g++) {
/* ms_mask can be used for other purposes in PNS and I/S,
* so must not apply M/S if any band uses either, even if
* ms_mask is set.
*/
if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
|| cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
|| cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
start += ics->swb_sizes[g];
continue;
}
for (i = 0; i < ics->swb_sizes[g]; i++) {
float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
float R = L - cpe->ch[1].coeffs[start+i];
cpe->ch[0].coeffs[start+i] = L;
cpe->ch[1].coeffs[start+i] = R;
}
start += ics->swb_sizes[g];
}
}
}
}
/**
* Encode scalefactor band coding type.
*/
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
{
int w;
if (s->coder->set_special_band_scalefactors)
s->coder->set_special_band_scalefactors(s, sce);
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
}
/**
* Encode scalefactors.
*/
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce)
{
int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
int off_is = 0, noise_flag = 1;
int i, w;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
for (i = 0; i < sce->ics.max_sfb; i++) {
if (!sce->zeroes[w*16 + i]) {
if (sce->band_type[w*16 + i] == NOISE_BT) {
diff = sce->sf_idx[w*16 + i] - off_pns;
off_pns = sce->sf_idx[w*16 + i];
if (noise_flag-- > 0) {
put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
continue;
}
} else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
sce->band_type[w*16 + i] == INTENSITY_BT2) {
diff = sce->sf_idx[w*16 + i] - off_is;
off_is = sce->sf_idx[w*16 + i];
} else {
diff = sce->sf_idx[w*16 + i] - off_sf;
off_sf = sce->sf_idx[w*16 + i];
}
diff += SCALE_DIFF_ZERO;
av_assert0(diff >= 0 && diff <= 120);
put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
}
}
}
}
/**
* Encode pulse data.
*/
static void encode_pulses(AACEncContext *s, Pulse *pulse)
{
int i;
put_bits(&s->pb, 1, !!pulse->num_pulse);
if (!pulse->num_pulse)
return;
put_bits(&s->pb, 2, pulse->num_pulse - 1);
put_bits(&s->pb, 6, pulse->start);
for (i = 0; i < pulse->num_pulse; i++) {
put_bits(&s->pb, 5, pulse->pos[i]);
put_bits(&s->pb, 4, pulse->amp[i]);
}
}
/**
* Encode spectral coefficients processed by psychoacoustic model.
*/
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
int start, i, w, w2;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = 0;
for (i = 0; i < sce->ics.max_sfb; i++) {
if (sce->zeroes[w*16 + i]) {
start += sce->ics.swb_sizes[i];
continue;
}
for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
s->coder->quantize_and_encode_band(s, &s->pb,
&sce->coeffs[start + w2*128],
NULL, sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
s->lambda,
sce->ics.window_clipping[w]);
}
start += sce->ics.swb_sizes[i];
}
}
}
/**
* Downscale spectral coefficients for near-clipping windows to avoid artifacts
*/
static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
{
int start, i, j, w;
if (sce->ics.clip_avoidance_factor < 1.0f) {
for (w = 0; w < sce->ics.num_windows; w++) {
start = 0;
for (i = 0; i < sce->ics.max_sfb; i++) {
float *swb_coeffs = &sce->coeffs[start + w*128];
for (j = 0; j < sce->ics.swb_sizes[i]; j++)
swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
start += sce->ics.swb_sizes[i];
}
}
}
}
/**
* Encode one channel of audio data.
*/
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce,
int common_window)
{
put_bits(&s->pb, 8, sce->sf_idx[0]);
if (!common_window) {
put_ics_info(s, &sce->ics);
if (s->coder->encode_main_pred)
s->coder->encode_main_pred(s, sce);
if (s->coder->encode_ltp_info)
s->coder->encode_ltp_info(s, sce, 0);
}
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
encode_pulses(s, &sce->pulse);
put_bits(&s->pb, 1, !!sce->tns.present);
if (s->coder->encode_tns_info)
s->coder->encode_tns_info(s, sce);
put_bits(&s->pb, 1, 0); //ssr
encode_spectral_coeffs(s, sce);
return 0;
}
/**
* Write some auxiliary information about the created AAC file.
*/
static void put_bitstream_info(AACEncContext *s, const char *name)
{
int i, namelen, padbits;
namelen = strlen(name) + 2;
put_bits(&s->pb, 3, TYPE_FIL);
put_bits(&s->pb, 4, FFMIN(namelen, 15));
if (namelen >= 15)
put_bits(&s->pb, 8, namelen - 14);
put_bits(&s->pb, 4, 0); //extension type - filler
padbits = -put_bits_count(&s->pb) & 7;
avpriv_align_put_bits(&s->pb);
for (i = 0; i < namelen - 2; i++)
put_bits(&s->pb, 8, name[i]);
put_bits(&s->pb, 12 - padbits, 0);
}
/*
* Copy input samples.
* Channels are reordered from libavcodec's default order to AAC order.
*/
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
{
int ch;
int end = 2048 + (frame ? frame->nb_samples : 0);
const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
/* copy and remap input samples */
for (ch = 0; ch < s->channels; ch++) {
/* copy last 1024 samples of previous frame to the start of the current frame */
memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
/* copy new samples and zero any remaining samples */
if (frame) {
memcpy(&s->planar_samples[ch][2048],
frame->extended_data[channel_map[ch]],
frame->nb_samples * sizeof(s->planar_samples[0][0]));
}
memset(&s->planar_samples[ch][end], 0,
(3072 - end) * sizeof(s->planar_samples[0][0]));
}
}
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AACEncContext *s = avctx->priv_data;
float **samples = s->planar_samples, *samples2, *la, *overlap;
ChannelElement *cpe;
SingleChannelElement *sce;
IndividualChannelStream *ics;
int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
int target_bits, rate_bits, too_many_bits, too_few_bits;
int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
int chan_el_counter[4];
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
/* add current frame to queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
} else {
if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
return 0;
}
copy_input_samples(s, frame);
if (s->psypp)
ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
if (!avctx->frame_number)
return 0;
start_ch = 0;
for (i = 0; i < s->chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
tag = s->chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
for (ch = 0; ch < chans; ch++) {
int k;
float clip_avoidance_factor;
sce = &cpe->ch[ch];
ics = &sce->ics;
s->cur_channel = start_ch + ch;
overlap = &samples[s->cur_channel][0];
samples2 = overlap + 1024;
la = samples2 + (448+64);
if (!frame)
la = NULL;
if (tag == TYPE_LFE) {
wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
wi[ch].window_shape = 0;
wi[ch].num_windows = 1;
wi[ch].grouping[0] = 1;
wi[ch].clipping[0] = 0;
/* Only the lowest 12 coefficients are used in a LFE channel.
* The expression below results in only the bottom 8 coefficients
* being used for 11.025kHz to 16kHz sample rates.
*/
ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
} else {
wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
ics->window_sequence[0]);
}
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = wi[ch].window_type[0];
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = wi[ch].window_shape;
ics->num_windows = wi[ch].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
ff_swb_offset_128 [s->samplerate_index]:
ff_swb_offset_1024[s->samplerate_index];
ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
ff_tns_max_bands_128 [s->samplerate_index]:
ff_tns_max_bands_1024[s->samplerate_index];
for (w = 0; w < ics->num_windows; w++)
ics->group_len[w] = wi[ch].grouping[w];
/* Calculate input sample maximums and evaluate clipping risk */
clip_avoidance_factor = 0.0f;
for (w = 0; w < ics->num_windows; w++) {
const float *wbuf = overlap + w * 128;
const int wlen = 2048 / ics->num_windows;
float max = 0;
int j;
/* mdct input is 2 * output */
for (j = 0; j < wlen; j++)
max = FFMAX(max, fabsf(wbuf[j]));
wi[ch].clipping[w] = max;
}
for (w = 0; w < ics->num_windows; w++) {
if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
ics->window_clipping[w] = 1;
clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
} else {
ics->window_clipping[w] = 0;
}
}
if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
} else {
ics->clip_avoidance_factor = 1.0f;
}
apply_window_and_mdct(s, sce, overlap);
if (s->options.ltp && s->coder->update_ltp) {
s->coder->update_ltp(s, sce);
apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
}
for (k = 0; k < 1024; k++) {
if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
return AVERROR(EINVAL);
}
}
avoid_clipping(s, sce);
}
start_ch += chans;
}
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
return ret;
frame_bits = its = 0;
do {
init_put_bits(&s->pb, avpkt->data, avpkt->size);
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
put_bitstream_info(s, LIBAVCODEC_IDENT);
start_ch = 0;
target_bits = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for (i = 0; i < s->chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
const float *coeffs[2];
tag = s->chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
cpe->common_window = 0;
memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
put_bits(&s->pb, 3, tag);
put_bits(&s->pb, 4, chan_el_counter[tag]++);
for (ch = 0; ch < chans; ch++) {
sce = &cpe->ch[ch];
coeffs[ch] = sce->coeffs;
sce->ics.predictor_present = 0;
sce->ics.ltp.present = 0;
memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
for (w = 0; w < 128; w++)
if (sce->band_type[w] > RESERVED_BT)
sce->band_type[w] = 0;
}
s->psy.bitres.alloc = -1;
s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
if (s->psy.bitres.alloc > 0) {
/* Lambda unused here on purpose, we need to take psy's unscaled allocation */
target_bits += s->psy.bitres.alloc
* (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
s->psy.bitres.alloc /= chans;
}
s->cur_type = tag;
for (ch = 0; ch < chans; ch++) {
s->cur_channel = start_ch + ch;
if (s->options.pns && s->coder->mark_pns)
s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
}
if (chans > 1
&& wi[0].window_type[0] == wi[1].window_type[0]
&& wi[0].window_shape == wi[1].window_shape) {
cpe->common_window = 1;
for (w = 0; w < wi[0].num_windows; w++) {
if (wi[0].grouping[w] != wi[1].grouping[w]) {
cpe->common_window = 0;
break;
}
}
}
for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
sce = &cpe->ch[ch];
s->cur_channel = start_ch + ch;
if (s->options.tns && s->coder->search_for_tns)
s->coder->search_for_tns(s, sce);
if (s->options.tns && s->coder->apply_tns_filt)
s->coder->apply_tns_filt(s, sce);
if (sce->tns.present)
tns_mode = 1;
if (s->options.pns && s->coder->search_for_pns)
s->coder->search_for_pns(s, avctx, sce);
}
s->cur_channel = start_ch;
if (s->options.intensity_stereo) { /* Intensity Stereo */
if (s->coder->search_for_is)
s->coder->search_for_is(s, avctx, cpe);
if (cpe->is_mode) is_mode = 1;
apply_intensity_stereo(cpe);
}
if (s->options.pred) { /* Prediction */
for (ch = 0; ch < chans; ch++) {
sce = &cpe->ch[ch];
s->cur_channel = start_ch + ch;
if (s->options.pred && s->coder->search_for_pred)
s->coder->search_for_pred(s, sce);
if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
}
if (s->coder->adjust_common_pred)
s->coder->adjust_common_pred(s, cpe);
for (ch = 0; ch < chans; ch++) {
sce = &cpe->ch[ch];
s->cur_channel = start_ch + ch;
if (s->options.pred && s->coder->apply_main_pred)
s->coder->apply_main_pred(s, sce);
}
s->cur_channel = start_ch;
}
if (s->options.mid_side) { /* Mid/Side stereo */
if (s->options.mid_side == -1 && s->coder->search_for_ms)
s->coder->search_for_ms(s, cpe);
else if (cpe->common_window)
memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
apply_mid_side_stereo(cpe);
}
adjust_frame_information(cpe, chans);
if (s->options.ltp) { /* LTP */
for (ch = 0; ch < chans; ch++) {
sce = &cpe->ch[ch];
s->cur_channel = start_ch + ch;
if (s->coder->search_for_ltp)
s->coder->search_for_ltp(s, sce, cpe->common_window);
if (sce->ics.ltp.present) pred_mode = 1;
}
s->cur_channel = start_ch;
if (s->coder->adjust_common_ltp)
s->coder->adjust_common_ltp(s, cpe);
}
if (chans == 2) {
put_bits(&s->pb, 1, cpe->common_window);
if (cpe->common_window) {
put_ics_info(s, &cpe->ch[0].ics);
if (s->coder->encode_main_pred)
s->coder->encode_main_pred(s, &cpe->ch[0]);
if (s->coder->encode_ltp_info)
s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
encode_ms_info(&s->pb, cpe);
if (cpe->ms_mode) ms_mode = 1;
}
}
for (ch = 0; ch < chans; ch++) {
s->cur_channel = start_ch + ch;
encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
}
start_ch += chans;
}
if (avctx->flags & CODEC_FLAG_QSCALE) {
/* When using a constant Q-scale, don't mess with lambda */
break;
}
/* rate control stuff
* allow between the nominal bitrate, and what psy's bit reservoir says to target
* but drift towards the nominal bitrate always
*/
frame_bits = put_bits_count(&s->pb);
rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
too_many_bits = FFMAX(target_bits, rate_bits);
too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
/* When using ABR, be strict (but only for increasing) */
too_few_bits = too_few_bits - too_few_bits/8;
too_many_bits = too_many_bits + too_many_bits/2;
if ( its == 0 /* for steady-state Q-scale tracking */
|| (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
|| frame_bits >= 6144 * s->channels - 3 )
{
float ratio = ((float)rate_bits) / frame_bits;
if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
/*
* This path is for steady-state Q-scale tracking
* When frame bits fall within the stable range, we still need to adjust
* lambda to maintain it like so in a stable fashion (large jumps in lambda
* create artifacts and should be avoided), but slowly
*/
ratio = sqrtf(sqrtf(ratio));
ratio = av_clipf(ratio, 0.9f, 1.1f);
} else {
/* Not so fast though */
ratio = sqrtf(ratio);
}
s->lambda = FFMIN(s->lambda * ratio, 65536.f);
/* Keep iterating if we must reduce and lambda is in the sky */
if (ratio > 0.9f && ratio < 1.1f) {
break;
} else {
if (is_mode || ms_mode || tns_mode || pred_mode) {
for (i = 0; i < s->chan_map[0]; i++) {
// Must restore coeffs
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
for (ch = 0; ch < chans; ch++)
memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
}
}
its++;
}
} else {
break;
}
} while (1);
if (s->options.ltp && s->coder->ltp_insert_new_frame)
s->coder->ltp_insert_new_frame(s);
put_bits(&s->pb, 3, TYPE_END);
flush_put_bits(&s->pb);
s->last_frame_pb_count = put_bits_count(&s->pb);
s->lambda_sum += s->lambda;
s->lambda_count++;
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = put_bits_count(&s->pb) >> 3;
*got_packet_ptr = 1;
return 0;
}
static av_cold int aac_encode_end(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
ff_psy_end(&s->psy);
ff_lpc_end(&s->lpc);
if (s->psypp)
ff_psy_preprocess_end(s->psypp);
av_freep(&s->buffer.samples);
av_freep(&s->cpe);
av_freep(&s->fdsp);
ff_af_queue_close(&s->afq);
return 0;
}
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
{
int ret = 0;
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
if (!s->fdsp)
return AVERROR(ENOMEM);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows(7);
if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
return ret;
if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
return ret;
return 0;
}
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
{
int ch;
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
for(ch = 0; ch < s->channels; ch++)
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
return 0;
alloc_fail:
return AVERROR(ENOMEM);
}
static av_cold void aac_encode_init_tables(void)
{
ff_aac_tableinit();
}
static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
int i, ret = 0;
const uint8_t *sizes[2];
uint8_t grouping[AAC_MAX_CHANNELS];
int lengths[2];
/* Constants */
s->last_frame_pb_count = 0;
avctx->extradata_size = 5;
avctx->frame_size = 1024;
avctx->initial_padding = 1024;
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
/* Channel map and unspecified bitrate guessing */
s->channels = avctx->channels;
ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
"Unsupported number of channels: %d\n", s->channels);
s->chan_map = aac_chan_configs[s->channels-1];
if (!avctx->bit_rate) {
for (i = 1; i <= s->chan_map[0]; i++) {
avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
69000 ; /* SCE */
}
}
/* Samplerate */
for (i = 0; i < 16; i++)
if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
break;
s->samplerate_index = i;
ERROR_IF(s->samplerate_index == 16 ||
s->samplerate_index >= ff_aac_swb_size_1024_len ||
s->samplerate_index >= ff_aac_swb_size_128_len,
"Unsupported sample rate %d\n", avctx->sample_rate);
/* Bitrate limiting */
WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
"Too many bits %f > %d per frame requested, clamping to max\n",
1024.0 * avctx->bit_rate / avctx->sample_rate,
6144 * s->channels);
avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
avctx->bit_rate);
/* Profile and option setting */
avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
avctx->profile;
for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
if (avctx->profile == aacenc_profiles[i])
break;
if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
avctx->profile = FF_PROFILE_AAC_LOW;
ERROR_IF(s->options.pred,
"Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
ERROR_IF(s->options.ltp,
"LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
WARN_IF(s->options.pns,
"PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
s->options.pns = 0;
} else if (avctx->profile == FF_PROFILE_AAC_LTP) {
s->options.ltp = 1;
ERROR_IF(s->options.pred,
"Main prediction unavailable in the \"aac_ltp\" profile\n");
} else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
s->options.pred = 1;
ERROR_IF(s->options.ltp,
"LTP prediction unavailable in the \"aac_main\" profile\n");
} else if (s->options.ltp) {
avctx->profile = FF_PROFILE_AAC_LTP;
WARN_IF(1,
"Chainging profile to \"aac_ltp\"\n");
ERROR_IF(s->options.pred,
"Main prediction unavailable in the \"aac_ltp\" profile\n");
} else if (s->options.pred) {
avctx->profile = FF_PROFILE_AAC_MAIN;
WARN_IF(1,
"Chainging profile to \"aac_main\"\n");
ERROR_IF(s->options.ltp,
"LTP prediction unavailable in the \"aac_main\" profile\n");
}
s->profile = avctx->profile;
/* Coder limitations */
s->coder = &ff_aac_coders[s->options.coder];
if (s->options.coder == AAC_CODER_ANMR) {
ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
"The ANMR coder is considered experimental, add -strict -2 to enable!\n");
s->options.intensity_stereo = 0;
s->options.pns = 0;
}
ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
"The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
/* M/S introduces horrible artifacts with multichannel files, this is temporary */
if (s->channels > 3)
s->options.mid_side = 0;
if ((ret = dsp_init(avctx, s)) < 0)
goto fail;
if ((ret = alloc_buffers(avctx, s)) < 0)
goto fail;
put_audio_specific_config(avctx);
sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
for (i = 0; i < s->chan_map[0]; i++)
grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
s->chan_map[0], grouping)) < 0)
goto fail;
s->psypp = ff_psy_preprocess_init(avctx);
ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
s->random_state = 0x1f2e3d4c;
s->abs_pow34 = abs_pow34_v;
s->quant_bands = quantize_bands;
if (ARCH_X86)
ff_aac_dsp_init_x86(s);
if (HAVE_MIPSDSP)
ff_aac_coder_init_mips(s);
if ((ret = ff_thread_once(&aac_table_init, &aac_encode_init_tables)) != 0)
return AVERROR_UNKNOWN;
ff_af_queue_init(avctx, &s->afq);
return 0;
fail:
aac_encode_end(avctx);
return ret;
}
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption aacenc_options[] = {
{"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
{"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
{"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
{"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
{"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
{"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
{"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
{NULL}
};
static const AVClass aacenc_class = {
"AAC encoder",
av_default_item_name,
aacenc_options,
LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault aac_encode_defaults[] = {
{ "b", "0" },
{ NULL }
};
AVCodec ff_aac_encoder = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(AACEncContext),
.init = aac_encode_init,
.encode2 = aac_encode_frame,
.close = aac_encode_end,
.defaults = aac_encode_defaults,
.supported_samplerates = mpeg4audio_sample_rates,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &aacenc_class,
};
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